1. Field of the Invention
The present invention relates to a VoIP (Voice over Internet Protocol) system and a method for preventing data loss in the same.
2. Description of the Related Art
Most current systems for voice over an Internet Protocol (VoIP) are built as described in a block diagram (FIG. 1) which illustrates a construction of the VoIP system for voice communication, that is, placing and receiving internet-based calls in prior art.
As illustrated in FIG. 1, basically the conventional VoIP system for voice communication through internet comprises a transmission system 50 that converts the recipient's voice to voice data and transmits the voice data over the internet, and a receiving system 60 that receives the voice data from the transmission system and converts the voice data back to the voice. The transmission system 50 is equipped with a microphone 10, ADC (Analog to Digital Converter) 11, a voice encoder 12, and a transmitting protocol processing part 13. The receiving system 60 is equipped with a receiving protocol processing part 14, a voice decoder 15, DAC (Digital to Analog Converter) 16, and a speaker 17.
The conventional VoIP system is operated as follows:
The microphone 10 in the transmission system 50 creates an analogue voice signal from the inputted voice by the transmitter, and the ADC 11 converts the analogue voice signal to a digital voice signal.
The voice encoder 12 compresses the digital voice signal to generate compressed voice data, and the transmitting protocol processing part 13, in order to transmit the compressed voice data to the receiving part over the internet, attaches a header and a trailer to the compressed voice data and generates voice packets.
On the other hand, the receiving protocol processing part 14 in the receiving system 60 analyzes the voice packets received from the transmission part, and extracts the compressed voice data from the voice packets by removing the header and the trailer that are installed in the voice packets.
The voice decoder 15, through decompression of the compressed voice data, restores the digital voice signal, and the DAC 16 converts the digital voice signal to an analog voice signal.
The speaker 17 converts the analog voice signal to the voice of the transmitter to help the recipient to be able to listen to the transmitter's voice.
As explained above, voice communication quality over the VoIP—based system is generally dependent on the number of voice packets per time unit (i.e., compressed voice data generation rate) corresponding to the conversation speed of the transmitter as inputted through the microphone 10 in the transmission system 10.
In other words, in case the inputted conversation speed exceeds the allowable number of voice packets for transmission per time unit (i.e., Channel Capacity) over the internet, some of packets corresponding to the difference from the compression voice data generation rate and the channel capacity are not transmitted to the recipient and get lost, consequently deteriorating the speech quality.
Therefore, the transmitter is encouraged to adjust his or her conversation speed to be inputted through the microphone 10 in order to ensure that the voice packets he or she transmitted are safely transmitted to the recipient without a loss.
For example, since the compression voice data generation rate should not exceed the channel capacity assigned to himself or herself, the transmitter can temporarily stop voice input for a certain period of time and continue later for decreasing the compression voice data generation rate, in case the compression voice data generation rate exceeds the designated channel capacity to himself or herself.
Previously, the recipient, when there was a blank while listening to the transmitter's voice, let the transmitter know that he or she was having difficulty in catching the transmitter's voice, and the transmitter adjusted his or her voice volume accordingly.
However, the previous method described above had a problem because the conversation speed control was made primarily by the transmitter's sense, making it difficult to suppress the compressed voice data generation rate under the channel capacity. Thus, it was very hard to prevent the voice packet loss in the conventional system.